Streaming : Stream dropping with High bitrate

[Ilyas OUMNIH]

Hello,

I'm facing an issue when streaming with a high bitrate (between 50Mbts to 100Mbts, res:1080p/4K).

The stream keeps cutting and jumping frames even though the bitrate showing in the Janus interface is constant.

Do you have any idea why that is happening?

Best Regards,

Ilyas

[Lorenzo Miniero]

I don't think we ever tested such a bitrate for a single stream (especially considering they make no sense for "normal" WebRTC sessions), so it's very likely we're currently not equipped for that.

L.

[lizardpeter]

I'm having a similar issue. Extremely unfortunately, I seem to only be able to go up to around 25000 kbps at 720p and 20000 kbps at 1080p (at an absolute maximum) before the stream becomes stuttery. This is using VP8 from OBS WebRTC. This makes it almost useless for my use case. I've tried most of the fixes and none of them have improved much. I have not ruled out an encoder issue, so I have been in contact with the developers of OBS WebRTC and am waiting for their next release to see if the issue persists. If it does, this might be a Janus issue. My tests have been using the VideoRoom plugin and not the Streaming plugin.

I need to find a solution for streaming at something like 1080p 50,0000 kbps and 120 FPS (at a minimum).

[Lorenzo Miniero]

As I wrote above, I honestly doubt you can get those bitrates with how the code works right now. They're not a reasonable bitrate for regular WebRTC usage (which is the main target for Janus), so not a huge priority for me. You may have some more luck with the sendmmsg branch.

L.

[Lorenzo Miniero]

Notice that it's browsers themselves as well who may barf on such a high bitrate. You can try capturing what Janus sends (or tries to send) via unencrypted pcap:

https://www.meetecho.com/blog/capturing-webrtc-traffic-in-janus/

and then analyze that to see if you can see something wrong. The pcap2mjr tool can also be helpful to try and convert that to a playable  recording.

L.

[lizardpeter]

The sendmmsg branch did seem to slightly help with the issue. Shouldn't WebRTC be bitrate agnostic? I thought it was just designed to be a real-time protocol for sending video and audio. Until now, it does seem like WebRTC is mostly used for things like video calls and low-bitrate transmissions, but I don't see why supporting high bitrates wouldn't be part of support for WebRTC in general.